1. Field of the Invention
This invention relates generally to a compensation method and system for use in sonic transmission and reproduction systems and more particularly to a compensation method and system that uses parametric values to control or adjust processes having transforms or models with properties or responses like the components or elements used in the transmission or reproduction system.
2. Description of Related Art
Most audio reproduction systems use electromechanical loudspeakers to acoustically reproduce audio signals. The electrical, mechanical, and acoustical properties of the loudspeakers are often less than ideal, causing distortions, response anomalies, and other coloration of the sound. Many techniques are used to compensate for the loudspeaker's characteristics in order to improve perceived audio quality.
Functional or behavioral models of loudspeakers are used in practice and found in literature to develop such compensations. A good example of models and how the modeling process works is described in “Active Equalization of Loudspeakers”, Speaker Builder, February 1997. Models consolidate technical languages and are usually intended to imitate or simulate the acoustic responses of the speaker system from electrical stimulus. Model creation or synthesis frequently begins by making functional groupings of elements which collectively represent or behave like all or part of the speaker. Coil and magnet parts become motors, which are represented by resistors, inductors, capacitors, back EMF generators and other transformed parts. A combination of factors such as air volume, moving mass, acoustic loading, magnetic-braking, and mechanical losses might be analyzed and simplified to LCR resonator networks or circuits. Most often, the transformed electromechanical, acoustic, and mechanical representations expressed in the model are further simplified or reduced to fewer elements. The model still responds like the speaker, but the parts making the model no longer have exact behavioral equivalence to the parts making the speaker. Consequently, traditional models are neither intended, nor capable of making parametrically addressed zero-phase compensations when speaker parts are changed.
One could characterize and invert the frequency response, as well as other properties, of a well-conceived model and achieve linear-phase correction of the loudspeaker. The technique does work to a fashion, but its dedicated, inflexible circuitry or specialized process tied to the traditional model limits its use to a one-speaker design. Some high-quality crossover networks constructed to divide the signal spectrum amongst multiple drivers may have some conjugate response correction like this.
A low-frequency resonant boost is intentionally designed for most speakers. Frequently, traditional models are made to represent quantifiable and predictable acoustic behavior as well as other speaker design factors affecting bass response. Mechanical construction and properties of air determine frequency, resonant losses and the configuration's effect on acoustic output from the speaker. A good approximation to a zero-phase conjugate or same-order correction for a wider frequency range can be designed and implemented in this manner. Several components are needed to match the resonant behavior, but all interact with each other when adjustments are made for a different speaker of similar concept and design. Therefore, the operation is not strictly parametrically controlled, as the adjustments must be re-calculated from the model to create the minimum-phase or exact match needed for best fidelity with the new speaker. When more corrections are added the interaction problem becomes formidable. The system must be tuned experimentally or the model analyzed each time an adjustment is made. Consequently, lumped model processes for response flattening are inherently designed for a specific speaker. The process must be redesigned for other speakers.
Traditional curve-fitting methods can require hundreds of data points and corresponding adjustments to set up and many components or much processing power to match the acquired frequency response. Analog methods are impractical and digital processes require much computation and extensive architectures to do this. Neither can provide phase accurate responses or the hidden corrections described later without having knowledge of the speaker and its operation. Without a model, the effort to combine amplitude, time and phase corrections together from measured responses becomes formidable.
Some of the most important behaviors of loudspeakers (with respect to acoustically perceptible effect) cannot be modeled or implemented from traditional methods. Such behaviors include standing wave interference, modal breakup, and coupled resonance as well as nonlinear consequences from such potentially interacting acoustic and mechanical behaviors. Counterproductive random motion or breakup may occur. Even when the average response remains flat or is the same as other frequencies being reproduced, energy can build up during signal stimulus and be released when the signal changes or ceases. In addition, other spatial factors related to stiffness of moving parts and high frequency de-coupling for motions away from a driving voice coil need to be considered. Any of these can create source movements, delayed energy release, and phase error to binaural hearing. Often, such destructive responses can be invisible or very difficult to interpret from traditional microphone-and-spectrum-analyzer calibration methods.
For example, unwanted responses arising from nodal and standing wave behavior affect the settling time, directional behavior, and radiated output of a speaker. Frequently, these responses cause perceptual changes to intelligence signals yet may not be visible or recognizable from response plots. Mechanical motions having large stored energy can be out of phase at different parts of the transducer. The acoustic output might appear to be flat, but human binaural hearing can localize the behavior to its source and the altered perception can degrade stereo imaging.
Often, mechanical disturbances are audible yet invisible or hard to interpret from response measurements made using a frequency sweep and microphone. Parts of a radiating surface can vibrate with different phase relationships to other parts, so that their additive acoustic output is low compared to motions within the transducer and the energy storage involved. When signals at the node frequency change and suddenly stop the release of stored energy can interact with other signals at different frequencies. The resulting beat sounds between the two frequencies can be audible and very objectionable. Sounds with spectra in the interference frequency range may appear louder and granular. Human binaural hearing can localize the disturbance to the driver or surfaces from which directional lobes might bounce, thereby imparting further damage to the stage illusion from multi-speaker stereo reproduction. For this situation, frequencies creating the mechanical disturbance must be sufficiently attenuated to prevent unmasked reproduction of consequential responses. Experience has shown that a sharp, deep notch needed to do this removes enough energy around the correction frequencies to cause a nasal sound. If this inappropriate correction is modified to achieve flat response, then the mechanical sounds remain along with a potential undesirable balance aberration.
Many small loudspeakers are constructed with a transducer, enclosure, and some resonant means to extend bass response such as a port or passive radiator. Usually, these parts are designed to achieve a practical and economical compromise between efficiency, frequency response accuracy, bass extension, and acceptable distortion. Designers of inexpensive, low-powered systems generally opt for higher efficiency to reduce amplifier requirements along with related costs of power supplies and packaging. The compromise situation exposes many undesirable behavioral aspects.
Most traditional speaker correction methods apply some variation of amplitude equalization to flatten and extend response from speakers. Adjustments are sometimes done by ear. To be quantitative, one must acquire relevant data. The most common techniques to do this use spectral analysis from noise stimulus. Then, response plots or displays indicate how an equalizer is to be adjusted. More sophisticated techniques based on delayed acceptance or sampled windows can measure first-arrival responses from the speaker and remove higher-frequency room disturbance to create anechoic-like data. The intent is to capture information relevant either to a listener in a room or to standard measurement practice where a test microphone is usually specified and placed one meter from the speaker. Such technique creates a response that may sound balanced to the single-point test microphone. One or more known systems go slightly beyond this by adjusting path lengths, or time delays to align multiple speakers to a listening position.
Other techniques provide transient response waveforms, waterfall or successions of spectral plots after an event. Group delay and time-related information is acquired. Such data needs interpretation and has limited use for frequency response leveling practice. Some behavioral responses can be recognized but much more information must be known about the speaker. Measurement devices such as accelerometers, differential acoustic probes, as well as microphones, are needed for this. Instrumentation may be placed near a suspected behavior site and moved to explore how a response changes with position. Weighted notches can be tuned or slowly swept through suspected frequencies while subjectively observing noise production. More information is needed about dimensions of parts, listening positions, as well as floor, shelf, possibly a computer monitor, or other interceding objects that may be part of the listening environment. Other technical specifications or expressions are needed to complete the conjugate model capable of time-phase-accurate correction.
A human operator can assume an alignment role by adjusting a graphic equalizer, manually tuning a parametric filter, or changing settings to a crossover device. Commercial analog components perform these functions, but they have limitations. Graphic equalizers have up to 31 bands or resonators, parametric devices include several adjustable filters and a few have variable crossover and shelf functions. Many more filters are needed. Combinations of graphic and parametric equalizers are incapable of providing a large enough number of points, nor the exact phase and time response to effectively compensate complex behavior from a loudspeaker. Either the corrections do not match specific frequencies, thereby creating phase error, or the number of filters is inadequate to deal with settling time and standing wave issues. Group delay distortion, time-phase error, incomplete correction and other shortcomings are likely to outweigh other improvements.
DSP filters can create many more filter sections than is practical from analog circuits. Graphic equalizers made up with parametrically controlled sections have been used with specialized control-generating software to create room response leveling. Such processes are difficult to set up because the room interferes with the identification of important behavioral indicators. Without their input, conjugate response corrections are not possible. Standing-wave and nodal distortion corrections could be made from such a system. However, the awkward compiling and processing needed to parametrically move the compensated notches would be difficult. Most likely, a single point response pickup and FFT has been used for data input to the system. Such methods cannot respond to or provide the time-phase information needed to create a true conjugate response to the speaker. Analysis systems, such as MLSSA, can remove room interference from measurements, and can produce frequency, transient, and settling response data from a loudspeaker system. However, the large amount of data from these measurements must be interpreted. The multiple-band graphic equalizer is not a good choice to install the correction.
DSP systems can economically create many parametric filters and time-related processes that are impractical with analog circuitry. Traditional large-scale DSP systems have little means to identify and cull out speaker behavior from other measurement anomalies. Their frequency-domain responses are likely to add phase errors and to overlook delayed settling energy. The sound might improve for one listening position but it will degrade for all others. More likely, the reproduced sound will change without definitive improvement.
Hence, those concerned with the reproduction of sound have recognized the need for a system and method of modeling the complete behavior of sound reproduction devices such that conjugate responses to the sound reproduction device responses may be created. The need for a system and method employing modifiable conjugate response has also been recognized. Furthermore, the need has also been recognized for a system and method that compensates the reproduction of sound independent of the environment in which the sound is to be heard. The present invention fulfills these needs and others.